System for audio signal processing

ABSTRACT

A sound reproduction system comprising a digital audio signal input ( 1 ), a digital audio signal processor ( 2 , DSP) and a digital audio signal output ( 3 ) wherein the digital signal processor ( 2 , DSP) comprises a high pass (HP) filter ( 21 ) with a high pass frequency (f), an amplifier ( 22 ) for a signal filtered by the HP filter, and a low pass (LP) filter ( 23 ) with a low pass frequency (f) for filtering the signal after amplification by the amplifier ( 22 ) and for providing an output signal, and the digital processor comprises an establisher ( 24, 25 ) for establishing the high pass frequency or the low pass frequency and a matcher ( 26 ) for matching the high pass frequency and low pass frequency of the high pass filter and low pass filter respectively to each other.

This invention relates to the field of sound reproduction, and moreparticularly to the field of digital audio signal processing.

The invention relates to a sound reproduction system comprising adigital audio signal input, a digital audio signal processor and adigital audio signal output.

The invention also relates to an audio signal processor for processingan incoming audio signal in an audio output signal. In particular theinvention relates to a digital signal processor (DSP) circuit orprogram.

The invention also relates to a method for processing a digital audiosignal.

A sound reproduction system, such as e.g. a loudspeaker telephonesystem, includes an output transducer, often called a loudspeaker, andan input for an audio signal. The loudspeaker produces sound pressurewaves in response to the audio input signal which is representative of adesired sound pressure wave.

Intelligibility of the sound as perceived by the listener is veryimportant, especially in noisy environments. The simplest way toincrease the intelligibility is to increase the average SPL (Soundpressure level), i.e. turning up the volume. However, simply turning upthe volume does not always lead to a more intelligible sound. Also, toohigh an output may lead to an overloading of a loudspeaker resulting ina further decrease of the intelligibility.

A number of attempts have been made to increase the intelligibility ofsound

United States Patent application US 2002/0015503 proposes e.g. toincrease intelligibility by individually constraining the gain factorsfor different frequency bands.

U.S. Pat. No. 6,011,853 describes a system in which the frequencyspectrum of the noise is measured and the speech signal is equalized forthe effect of noise at a particular frequency.

The existing systems and methods are, however, either very complicated,requiring complicated computations and thereby complicated circuitry(hard-ware), or, in the case of a program (soft-ware) being used, acomplex program, or supply only a limited advantage.

Notwithstanding the above-mentioned reference, there continues to exista need in the art for improved systems and methods providing improvedintelligibility.

It is an object of the present invention to provide a sound reproductionsystem and method with improved intelligibility.

To this end, the sound reproduction system in accordance with theinvention is characterized in that the system comprises a digital signalprocessor, the digital signal processor comprising a high pass (HP)filter with a high pass frequency, an amplifier for amplification of asignal filtered by the HP filter, and a low pass (LP) filter forfiltering the signal after amplification and for providing an outputsignal, and the digital processor comprises an establisher forestablishing the high pass frequency or the low pass frequency and amatcher for matching the high pass frequency and low pass frequency ofthe high pass filter and low pass filter respectively to each other.

The system in accordance with the invention is based on the followinginsights:

-   1. The incoming signal is amplified (by the amplifier), to increase    the loudness, however-   2. increasing the incoming signal could lead to a signal higher than    the maximum digital signal, in such circumstances the signal would    often be digitally clipped, leading to a distortion of the signal.-   3. Low frequencies are removed from the signal prior to    amplification, this allows the remainder of the signal to be    amplified with, on average, a higher gain factor. This is done by    the HP filter, situated before amplification. The lower frequencies    are, in so far as intelligibility is concerned, of relatively low    importance. The lower frequencies comprise much of the amplitude of    the signal, so removing the lower frequencies strongly reduces the    amplitude of the signal, creating headroom for amplification, i.e. a    stronger amplification for the remainder of the signal. A large part    of the amplitude of especially speech is comprised in the lower    frequencies so attenuating lower frequencies allows for a    considerable increase in head room (i.e. amplification without    hitting a clipping level).-   4. Simply cutting the lower frequencies and increasing the output    would lead to an increase in intelligibility, however it does not    always lead to a readily perceivable increase in intelligibility.    Due to the use of the high pass filter, the signal comprises a    relatively high proportion of high frequency tones leading to a    perceptually annoying shrill signal, a discoloration of the audio    signal, reducing the intelligibility. Furthermore the amplification    may lead to the introduction of the overtones. The low pass (LP)    filter, after amplification, restores the balance, and in addition    may cut out or at least reduce the overtones produced by the    amplifier leading to a more natural sound, reducing at least in part    the discoloration of the signal and increasing the intelligibility.-   5. Within the system in accordance with the invention the cut-off    frequencies are matched, i.e. that there is a relationship between    the established HP cut-off frequency and the LP cut-off frequency,    wherein the value of the low pass frequency and/or the high pass    frequency are matched in opposite directions, meaning that if the HP    cut-off frequency is reduced (lowered) the LP cut-off frequency is    increased and vice versa, or to put it differently the cut-off    action of the HP filter and LP filter are matched, in that if one    action is large, the other is too, and if one action is relatively    small, the other is too. The HP and LP filters thus are used as    coupled elements in the speech or sound processing, they are not    elements with unrelated parameters. The cut-off action of the    high-pass filter provides for a discoloration of the audio signal,    and the LP filter's cut-off action is matched to counteract this    effect. The system has a means for establishing either the HP or LP    cut-off frequency and matching the LP or HP cut-off frequency to the    established frequency.

There are several preferred embodiments:

In embodiments the system comprises a sensor for measuring back-groundnoise level, and comprises an element having as an input the measurednoise level and as an output the HP cut-off frequency, wherein the HPcut-off frequency increases as the background noise level increases, andthe LP cut-off frequency decreases as the HP cut-off frequencyincreases. This decrease of the LP cut-off frequency may be performed bychanging (decreasing) the cut-off frequency of a single LP filter as afunction of the HP cut-off frequency, or, alternatively, the system maycomprises a set of different LP filters, and dependent on the HP cut-offfrequency the signal, after amplification is directed to one of the setof LP filters. The latter embodiment is particular useful if morecomplex LP filters are used, having more than one variable, i.e. moreparameters being changed than just the LP cut-off frequency. Thisselection may be performed with a look up table wherein dependent on theHP cut-off frequency the signal undergoes different digital calculationsequivalent to different LP filters. This may be performed in two steps,in which in a first step the noise level is determined, this triggers achoice of the HP cut-off frequency, and then from the HP cut-offfrequency the LP cut-off frequency is chosen.

Preferably the system, and/or program comprises a means for establishingthe cut-off frequency of the high pass filter in dependence on theaverage amplification in the amplification stage. The averageamplification is a measure for the average gain of the signal andthereby of the loudness level of the emitted sound signal. It isadvantageous when the HP cut-off frequency of the high-pass filterincreases with increasing average loudness level of the emitted signal,and in addition the cut-off frequency of the low pass filter is matched,i.e. decreases in step with the changing HP cutoff frequency. At veryhigh amplification levels (as could happen when the system is used inloud, noisy environments) there is a large need for the filtering actionof the HP filter which introduces a relatively large unbalance in soundand in addition the large amplification level itself may introduce arelatively large distortion of the signal, wherein the unwanted,unnatural overtones of the signal after amplification comprise muchenergy. This leads to a harsh sound. The harshness of the soundsometimes and even often, as the inventors have realized leads to thelistener keeping the loudspeaker, especially in mobile telephones, atsome distance from the ear. Apart from the fact that holding theloudspeaker at some distance from the ear in itself will lead to aconsiderable reduction in signal to noise, since the signal will bereduced and the noise will be increased, the mere fact that the sounditself is perceived as harsh in fact means a reduction inintelligibility of the message given. For vocal messages the harshnessof the voice often is an integral part of the message which the personspeaking wishes to address the listener, sometimes even more importantthan the actual words of the messages. It is thus important for theintelligibility of the message, seen in a concept broader than merelywhether or not words are understood, that a clear “natural” voicetransfer is achieved. At lower amplification levels the “harsh sound”effect, i.e. the discoloration of the signal is much less audible. Inshort, at high amplification levels there is a relatively largeunbalance in the signal (much more energy in high frequencies) and inaddition a relatively large part of the amplitude at high frequencies isdue to artificial artifacts due to the amplification, and removing (intoto or partially) the higher frequencies removes the unbalance andartifacts thus leading to a more natural sound, whereas at relativelylow amplification levels, considerably more of the signal amplitude athigh frequencies is of natural origin and much less due to artificialartifacts, so setting the cut-off frequency at a relatively highfrequency is then preferable. By establishing the cut-off frequency ofthe high pass filter dependent on the amplification and simultaneouslymatching the cut-off frequency of the low pass filter an improved soundreproduction is possible. To some extent this embodiment aims to have asimilar effect as the first mentioned embodiment with a noise sensor,however, instead of coupling the matched cut-off action of HP and LPfilter to a measured noise level, the matched LP and HP filter cut-offaction is coupled to an amplification level. Usually the higheramplification levels will be used in high background noise levels, (auser will turn up the volume in noisy environments) so in a sense theuser then acts as a noise sensor. Using a noise sensor as a part of thesystem is from a technical point of view preferred, but will increasethe cost and complexity of the system.

In embodiments the system comprises means to measure/establish thesample frequency fs of the incoming signal and set the maximum cut-offfrequency of the LP filter at fs/2. Any signal above fs/2 is not acomponent of the original signal, but due to overtones. The samplefrequency fs thus in fact determines the maximum cut-off frequency ofthe LP filter. Therefore in these embodiments the cut-off frequency hasan upper limit of fs/2. The sample frequency fs may be determined by thebandwidth of the incoming signal. For instance for Internet audio-videothe bandwidth is often rather small. Also, if the audio system itselfhas a rather low power, there is not much use in having a very largebandwidth. So, in preferred embodiments the system has means toestablish the sample frequency of the signal, which may be establishedfrom the incoming signal, or may be established as a function of thepower restriction of the audio system (for instance if the system may behooked up to different physical amplifiers and thus power restrictionsmay apply). For high values of fs, for instance 44 kHz, this does nothave much influence on the cut-off action of the low pass filter,however this is not the case for low values of fs (such as for instance8 kHz, 11.025 kHz, 16 kHz, 22.05 kHz fs/2 is rather small 4 kHz, 5.5kHz, 8 kHz respectively 11 kHz). This restriction however ‘colors’ thesound, since frequencies above fs/2 are absent, in fact the originalsignal as received is already ‘discolored’. The system in the preferredembodiment has a means for determining the HP cut-off frequency independence of the established sample frequency fs to counteract thediscoloration. Again, this may be done in several manners: The increaseof the HP cut-off frequency may be performed by changing (increasing)the cut-off frequency of a single HP filter as a function of the LPcut-off frequency, or, alternatively, the system may comprises a set ofdifferent HP filters, and dependent on the cut-off frequency the signalis lead to one of the set of HP filters. The latter embodiment isparticular useful if more complex filters are used, having more than onevariable, i.e. more parameters being changed than just the cut-offfrequency. This selection may be performed with a look up table whereindependent on the LP cut-off frequency the signal undergoes differentdigital calculations equivalent to different HP filters.

In most preferred embodiments the system combines both of the preferredembodiments, i.e. it comprises means for, on the one hand, establishingthe sample frequency fs and then determining the HP cut-off frequency onthe basis of the fs value, and on the other hand, measuring the noiselevel and determining the HP cut-off filter value and then determiningthe LP cut-off filter frequency.

Preferably the high pass frequency lies between 300 and 2 kHz, the lowpass frequency lies above 2 kHz and fs/2 where fs is the samplefrequency.

Preferably the amplifier is arranged not to amplify a signal having asignal strength below a threshold value.

Below a threshold value (a certain minimum amplitude) the signal isprobably due to noise. Not amplifying such signals improvesintelligibility, since the noise is reduced. Furthermore the differencebetween silence and speech is better distinguishable, which alsoincreases intelligibility. The threshold may be set at initializationtime or at run time, in case the user needs to set the threshold value,e.g. when making several phone calls, since the signal noise can dependon the GSM provider or the GSM device used by the far end user.

The high pass filter is preferably a first order or second order filter,i.e. a filter with a relatively gradual slope. It is advantageous toremove much of the energy of the low frequency components of theincoming signal to provide head room for the amplification. However, afilter with a slope that is relatively steep (a step filter being themost extreme example of such a filter) removes so much of the lowerfrequency components that this may result in an unnatural soundingvoice. Preferably the system comprises a means for enabling a user tochange the order and/or the cut/off frequency. Using a 2^(nd) order highpass filter results in good speech intelligibility and/or signalloudness, whereas using a 1^(st) order high pass filter will preservethe more natural sound of the original signal.

In a preferred embodiment the system comprises the high pass (HP) filterfollowed by an AGC followed by a limiter/clipper followed by the lowpass (LP) filter. This embodiment is preferred when in circumstanceswhere signal loudness is of prime interest. A limiter scans for peaks inthe audio signal and attenuates the audio portion around the peak if theattenuation is necessary to limit the amount of clipping, while yet forvery loud signals allowing for clipping.

In a different preferred embodiment, the system comprises an automaticvolume leveler preceded or preferably followed by the high pass (HP)filter, providing a leveled signal, followed by a gain, and a clipper,followed by the low pass (LP) filter. This embodiment is preferred whenlow computational effort is preferred

(Hard) clipping is a simple operation in which any signal above athreshold signal strength is reduced to said given threshold signalstrength, i.e. a maximum signal strength is set. The advantage of suchan embodiment is that a simple system is used, the disadvantage is thatthe signal is more heavily distorted, since any details in the signalabove the threshold signal are lost.

In all embodiment the HP and LP cut-off frequencies are coupled. Thisforms the core of the invention, the HP and LP filter actions arematched wherein the discoloration due to the application of the onefilter is at least partly compensated by the action of the other filter.Often the ‘leading filter’ will be the HP filter, i.e. the HP cut-offfrequency will be established on the basis of a parameter and the LPcut-off frequency will follow, but in embodiments wherein the incomingsignal has a small sample frequency the ‘leading filter’ may be the LPfilter.

In preferred embodiments the system comprises a measuring system, suchas a microphone, for measuring background noise levels.

Preferably for one or more of the parameters the dependency on themeasured noise level is non-linear.

Within the concept of the invention a ‘clipper’, ‘compressor’,‘amplifier’, ‘filter’, ‘converter’, ‘comparator’ etc are to be broadlyunderstood and to comprise e.g. any piece of hard-ware (such as aclipper, compressor, amplifier etc), any circuit or sub-circuit designedfor performing a clipping, compression, amplification etc function asdescribed as well as any piece of soft-ware (computer program or subprogram or set of computer programs, or program code(s)) designed orprogrammed to perform a clipping, compressing, filtering etc operationin accordance with the invention as well as any combination of pieces ofhardware and software acting as such, alone or in combination, withoutbeing restricted to the below given exemplary embodiments. One programmay combine several functions.

The invention is also embodied in any computer program comprisingprogram code means for performing a method in accordance with theinvention when said program is run on a computer as well as in anycomputer program product comprising program code means stored on acomputer readable medium for performing a method in accordance with theinvention when said program is run on a computer, as well as any programproduct comprising program code means for use in a telephone system inaccordance with the invention, for performing the action specific forthe invention.

These and further aspects of the invention will be explained in greaterdetail by way of example and with reference to the accompanyingdrawings, in which

FIG. 1 is a schematic diagram of a system including a loudspeaker, and aDSP.

FIG. 2 shows schematically a DSP in accordance with the invention

FIG. 3 illustrates schematically matching of LP filter frequency to HPfilter frequency

FIG. 4 illustrates schematically an alternative manner of matching LPfilter frequency to HP filter frequency

FIG. 5 illustrates schematically matching of HP filter frequency to LPfilter frequency

FIG. 6 illustrates schematically an alternative manner of matching HPfilter frequency to LP filter frequency

FIG. 7 illustrates matching of LP and HP filter frequency in a graphicalform.

FIG. 8 shows two examples of high pass filters which may be used in theinvention.

FIG. 9 illustrates one type of embodiments of the invention.

FIG. 10 illustrates a different type of embodiment.

FIG. 11 illustrates the AVL behavior of an AVL.

FIG. 12 illustrates a preferred embodiment in which parameters areadapted in dependence on a measured noise level.

The present invention will now be described more fully hereinafter withreference to the accompanying drawings, in which preferred embodimentsof the present invention are shown. This invention may, however, beembodied in many different forms and should not be construed as limitedto the embodiments set forth herein. Like numbers refer to like elementsthroughout.

FIG. 1 illustrates schematically a sound reproduction system. Suchsystem can for instance be a hands-free loudspeaker cellularradiotelephone for use in an automobile. When implemented as ahands-free cellular telephone, speech signals received from a far end,i.e. from a distant party, are transmitted from a cellular base station(not shown), received by the transceiver of the cellular phone (notshown), and applied to the input 1 for an incoming far end signal as aninput waveform W. In this example it is assumed that the transmissionback and forth between the system, in this example a telephone system,and the far end is in a digital form. If the original signals are in ananalog form the system comprises an A-to-D converter to generate adigital far end signal which is then fed into input 1.

As shown in FIG. 1, the waveform is applied in a digital format at input1 of or connected to a DSP (digital sound processor) 2, which isconnected to or which comprises a digital output 3. The digital signaloutput is fed to and converted to an analog format by D-to-A converter 4and amplified by amplifier 5 for use by the loudspeaker 6. A soundpressure wave W1 representative of the speech of the distant party isemitted by loudspeaker 5. Accordingly, the radiotelephone user hearssound pressure waveforms which are representative of the speech of thedistant party.

However, the listener does not just hear the sound generated by the loudspeaker, but also other sounds, which may make the sound generated bythe loud speaker difficult to understand, i.e. of low intelligibility.

Turning up the volume seems a first and obvious choice to increase theintelligibility. However, the maximum output level of the loud speakeris often limited and simply turning up the volume often leads to morenoise, not necessarily a better intelligibility of the signal.

To improve the intelligibility a number of co-operating measures aretaken in the system and method in accordance with the invention.

FIG. 2 illustrates very schematically a DSP (digital sound processor)for use in a system in accordance with the invention. The DSP comprisesa high pass (HP) filter 21 with a cut-off frequency e.g. and preferablybetween 300 Hz and 2 kHz, e.g. between 500 and 1500, more preferablybetween 800 and 1200 Hz. The high pass filter removes or reducesfrequency components below the cut-off frequency f.

By the HP filter most of the energy of the signal has thereby beenremoved. This enables the remaining signal to be much more amplified(before running into problems in regards to a digital clipping, i.e. avalue higher than the maximum value) by amplifier 22. The HP filtering,however, increases the high frequencies relative to the low frequencies,leading a shrill sound. The application of a low pass filter (LP) 23after the amplification restores, at least in part the ‘naturalness’ ofthe sound. In the system in accordance with the invention the HP and LPcut-off frequencies f and f′ are matched, schematically indicated by thearrows between HP and LP filters. The system has a means forestablishing either one of the frequencies f or f′ and once one of thesefrequencies is established establishing the other frequency.

In the system in accordance with the invention the cut-off frequenciesand thereby the actions of the HP and LP filter are matched, i.e. theyare inter-correlated. As described to this end the system comprises onthe one hand an establisher 24, 25 to establish the HP or LP cut-offfrequency f or f′, and a matcher 26 to match the LP or HP cut-offfrequency to the established HP or LP cut-off frequency. In allembodiment the HP and LP cut-off frequencies are coupled. This forms thecore of the invention, the HP and LP filter actions are matched whereinthe discoloration due to the application of the one filter is at leastpartly compensated by the action of the other filter. “Matching” meansthat there is a relationship between f and f′, i.e. f=F(f′) or f′=F′(f).Often the ‘leading filter’ will be the HP filter, i.e. the HP cut-offfrequency will be established on the basis of a measured or determinedparameter and the LP cut-off frequency will follow, but in embodimentswherein the incoming signal has a small sample frequency the ‘leadingfilter’ may be the LP filter.

FIG. 3 illustrates schematically a high pass filter with a high passfilter frequency f, this filter frequency f may be influenced aparameter P, such as environment noise level, line noise level, aparameter set by the user, or the amplification. The establisherestablishes the HP cut-off frequency in dependence of such parameter orparameters. This influence is schematically indicated by the transversearrow through the frequency f and the arrow between P and f. The systemcomprises a matcher which matches the low pass frequency f′ of the lowpass filter 23 to the HP filter frequency f. “Matching” in the presentinvention should not be interpreted as meaning that f is made equal tof′ or vice versa, but that a relationship is established between thevalue f and the value f′, i.e. f′=F(f) where F stands for a function.This may be a fixed relationship, i.e. a fixed function F, or thefunction may be dependent itself on further parameters, which in FIG. 3is schematically shown by the letter P′ and the arrow between thisletter and the function F. Parameters may be for instance theenvironmental noise level N, line noise level, or the amplification(gain) of the amplifier in between the HP and LP filter. In such a casethe relation between f and f′ may be represented by f′=F(f, P′).

FIG. 4 illustrates a variation on the scheme shown in FIG. 3. In thisembodiment a number of LP filters are used, and, dependent on theestablished HP cut-off frequency f, and possibly additional parametersP′, the signal is, after amplification by amplifier 22, led, using thematcher, to one of the LP filters, each filter having a cut-offfrequency f′. This is schematically shown in FIG. 4 by the differentarrows interconnecting the HP filter and one of the LP filters, and bythe fact that the cut-off frequency is different in the different LPfilters.

Such an embodiment is more complicated then the embodiment in which asingle LP filter is used, however. The latter embodiment is particularuseful if more complex LP filters are used, having more than onevariable, i.e. more parameters being changed than just the LP cut-offfrequency, for instance also the order of the filter. This selection maybe performed with a look up table wherein dependent on the establishedHP cut-off frequency the signal undergoes different digital calculationsequivalent to different LP filters. This may be performed in two steps,in which in a first step the noise level is determined, this triggers achoice of the HP cut-off frequency, and then from the HP cut-offfrequency the LP cut-off frequency is chosen. The choice may bedependent on additional parameters.

FIG. 5 illustrates schematically an embodiment in which the LP filterfrequency is established, as a function of the sample frequency fs andthe HP filter cut-off is a function of the established LP filter cut-offfrequency f′, i.e. f=F(f′), the matcher 26 matches the frequencies fad nf, i.e. calculates f on the basis of the established frequency f′. Insome embodiments the function F itself is dependent on other parametersP′, such as e.g. the noise level.

FIG. 6 illustrates a variation of the embodiment illustrates in FIG. 5.In this embodiment a number of HP filters are used, and, dependent onthe established LP cut-off frequency f′, and possibly additionalparameters P′, the signal is, after amplification by amplifier 22, ledto one of the HP filters, each filter having a cut-off frequency f. Thisis schematically shown in FIG. 6 by the different arrows interconnectingthe LP filter and one of the HP filters, and by the fact that thecut-off frequency is different in the different HP filters.

Such an embodiment is more complicated then the embodiment in which asingle HP filter is used, however. The latter embodiment is particularuseful if more complex HP filters are used, having more than onevariable, i.e. more parameters being changed than just the HP cut-offfrequency, for instance also the order of the filter. This selection maybe performed with a look up table wherein dependent on the establishedLP cut-off frequency the signal undergoes different digital calculationsequivalent to different HP filters. This may be performed in two steps,in which in a first step the sample frequency fs is determined, thistriggers a choice of the LP cut-off frequency, and then from the LPcut-off frequency the HP cut-off frequency is chosen. The choice may bedependent on additional parameters.

FIG. 7 illustrates in a graphical form the relation between the LPcut-off frequency f′ and the HP cut-off frequency f. The relation isfound by drawing a vertical line for a particular value of theparameters (e.g. the noise) between the HP (f) and LP (f′) cut-offfrequency graphs, and reading the corresponding curve values (indicatedby the double-headed arrow). As shown schematically, the graphs may becontinuous (f′₁(P)_(LP) and f₁(P)_(HP)), or stepped (f′₂(P)_(LP) andf₂(P)_(HP)), as may be for instance the case when a number of distinctLP of HP filters are used. The curves themselves may be dependent onparameters P and/or P′, such as for instance a measured noise level N,the line noise level, the chosen amplification level (when such may beset by the user), the same frequency fs.

The amplification is done by an amplifier 22 (see FIG. 2).

This amplifier 22 is preferably a compressing amplifier. A compressingamplifier is an amplifier which amplifies the signal but also levels theaverage sound level, i.e. sounds having a small amplitude are moreamplified than sound having a high sound level, thus reducing the signalamplitude range. This may be done in several manners e.g. aclipper/limiter arrangement, a clipper/compressor or an AVL (automatedvoltage leveler) followed by a gain and a clipper. A number of differenttechniques may be used, including using look-up tables to perform theamplification and compression. The amplitude range, in particular theupper limit of the range may be set by the manufacturer or influenced bythe user e.g. by means of a loudness setter (a knob with which the usermay set the loudness). Compared to a straight forward, linearamplification of the signal (i.e. for all sound levels an equalamplification factor) a compressed amplification leads to moreintelligibility of the sound, especially in case the environmental noiseis high, causing the silent voice parts to be masked by the backgroundnoise. The words are more easily distinguishable, and thus theintelligibility of the sound is improved.

It could also lead, however, to a distortion of the sound since thenon-linear amplification of the sound introduced overtones (highfrequency components at double, triple etc the original frequency) whichleads to an increased harshness of the sound. This is perceived by thelistener as being unpleasant, and in fact, in an important sense,reduces the intelligibility, in a broader sense, of the vocal message,since the harshness of the spoken words often forms an important aspectof the vocal message. This effect is present even without clipping,although the clipping itself also introduces overtones, even moredistorting the signal.

Intelligibility in a broader sense does not just relate to the words assuch, but also to the message the speaker wishes to convey to thelistener. The harshness of the sound, especially at higher averageamplification, makes everybody sound angry, thus strongly reducing thefinesses in emotions the speaker wishes to convey.

The application of a low pass filter, as in the inventive system,depicted in FIG. 2, after the compressing amplifier reduces theperceived harshness of the voice, restoring at least to some extent theoriginal emotional content of the spoken words, i.e. giving a much morenatural sound.

It is remarked that in most western languages the pitch of a wordinfluences the emotional impact of the word, but not the meaning of theword per se. However, there are languages in which the pitch of the wordplays a much larger role, leading to completely different meanings forone and the same “word” depending on the pitch of the word. When suchlanguages are used (which cannot be excluded) the use of the low passfilter becomes even more advantageous. The invention is in particular ofadvantage when used in conjunction with or in automated speechrecognition systems, especially for languages in which the pitch of thespoken word influences the meaning of the words. What is above discussedin relation to spoken words, i.e. voice, equally applies when the soundreproduction system is used to reproduces music. Also in music the waymusic is perceived is of course dependent on whether one can hear thenotes, but also the harshness of the sound is very important. Theinvention is thus, although of great importance to systems in whichvocal messages are relayed such as telephone systems, not restricted tosuch systems, systems for reproducing music may equally benefit of thisinvention.

A leveling or compression action may be performed before HP filtering ofthe incoming signal or after it has been HP filtered. The amplificationper se (i.e. the gain) is done after HP filtering. The clipping is doneafter the gain or in conjunction with the gain. When use is made of aclipper the low pass filter is positioned after the clipper.

As explained above the relation between HP and LP cut-off frequency maybe fixed or dependent on a number of parameters.

The amplification stage and/or the clipping stage may also lead to adiscoloration of the signal, in more sophisticated embodiments of thesystem the matcher for matching the HP and LP frequency comprises aninput for at least one parameter of the element between the HP and LPfilter, such as the amplifier and/or clipper, whereby the relationbetween the HP and LP cut-off frequency is dependent on saidparameter(s), i.e. f′=F(f,P) of f=F′(f′,P′) where P and P′ stands forthe parameter(s) of such intermediate elements. For instancenon-linearities introduced by a clipper and/or a leveling amplifier mayintroduce higher overtones, so dependent on the presence or theeffectiveness of such elements the LP filter cut-off action of the LPfilter may be increased.

FIG. 8 gives two examples of high pass filters usable in a system inaccordance with the invention.

The left hand side of the figure illustrates a 1^(st) order filter, theright hand side a second order filter. The shown high pass filter have acut-off frequency f of approximately 1 kHz. First or second orderhigh-pass filters (which have relatively moderate sloops of 5-15 dB peroctave) are preferred. Removing too much of the low frequenciescomponent results in a very unnatural sounding voice (or unnatural oddsounding music) Therefore the order of the high pass filter ispreferably limited at 2. This also reduces the computational powerrequired. Preferably the user can change the high pass filter from1^(st) to 2^(nd) order and vice versa, or the system comprises anautomatic switching mechanism dependent on the incoming signal. Usingthe 2^(nd) order results in high speech intelligibility (in restrictedsense, i.e. only the words) and/or signal loudness, whereas the 1^(st)order HP filter will better preserve the natural sound of the originalsignal.

The HP filter may for instance consist of a biquad whose coefficientsare listed in Table 1, according to the format

TABLE 1${H(z)} = {\frac{b_{0} + {b_{1}z^{- 1}} + {b_{2}z^{- 2}}}{1 + {a_{1}z^{- 1}} + {a_{2}z^{- 2}}}.}$Filter coefficients of the HP filters depicted in FIG. 3. 1^(st) order2^(nd) order b₀ 0.70710678118655 0.56903559372885 b₁ −0.70710678118655−1.13807118745770 b₂ 0.00000000000000 0.56903559372885 a₁−0.41421356237310 −0.94280904158206 a₂ 0.00000000000000 0.33333333333333

The lower frequencies contribute mainly to the specific sound of thevoice, but less to the speech intelligibility. This property forms oneaspect of the invention.

By attenuating the lower frequencies, the signal amplitude will decreasesignificantly creating headroom to amplify the remaining signal whichcontains relatively more frequencies contributing to the speechintelligibility.

When amplifying the speech signal afterwards, even when compressing andclipping it, the speech intelligibility will be better than without theuse of the HP filter for mainly two reasons:

-   -   the signal contains relatively more frequencies contributing to        the speech intelligibility    -   the low frequencies are less hard clipped, resulting in less        harmonics (due to clipping) disturbing the speech        intelligibility.

Removing too much of the low frequencies, however, would results in avery unnatural sounding voice. Therefore, the HP filter is preferablyonly first order e.g. a Butterworth (first order IIR) filter. This hadthe advantage of little computational power consumption.

FIG. 9 illustrates a detail on a system in accordance with an embodimentof the invention.

The DSP comprises a HP filter (for instance one as illustrated in FIG.8, in this example for instance a Butterworth 1^(st) or 2^(nd) orderfilter with a cut-off frequency value of for instance between 50 and 2kHz), followed by an AGC, followed by a limiter/clipper, followed by alow pass filter (LP), with a cut-off frequency between 2 kHz and fs/2.

In this example all audio streams may be mono. The sample rate frequencycan be e.g. one of the following: 8 kHz, 11.025 kHz, 16 kHz, 22.05 kHz,32 kHz, 44.1 kHz or 48 kHz

For an incoming signal with a very low sample frequency, for instance of8 kHz, the LP frequency cut-off frequency is set at fs/2, i.e. at 4 kHz,the HP cut-off frequency is then set at for instance 400 Hz, for asample frequency of 16 kHz the HP cut-off frequency is set at 200 Hz,and so forth. So the sample frequency fs then establishes the LP cut-offfrequency, and the HP cut-off frequency is matched to the LP cut-offfrequency.

In embodiments the AGC (Automatic Gain Control) acts block based,meaning that the gain factor only changes per block. In this way thecomputational power is kept to a minimum.

The gain may be calculated for instance as follows:

First of all, the running RMS (Root Mean Square) value of the inputsignal is computed. This RMS value is a smoothed average based on therecent “history” of the signal waveform. Then the peak value iscalculated using a look-ahead time in order to anticipate to upcomingsignal peaks.

With the RMS and the peak value, the crest-factor is computed. Aso-called “depeak” factor is used to specify how hard the algorithm canclip peaky signals (high values will yield more clipping). Afterwardsthe gain is computed and is compared with the maximal allowed gain,which can be set by the user, and the minimal value of the two ischosen. Although not shown here the maximum allowed gain setting can bean input for the high pass filter, wherein the cut-off frequency is afunction (or more in general one or more characteristics of the filter,which could apart from the value of the cut-off frequency also oralternatively e.g. include switching from a first order to a secondorder) of the maximum allowed gain setting.

The maximum allowed gain setting can be an input for the high passfilter, wherein the cut-off frequency is a function (or more in generalone or more characteristics of the filter, which could apart from thevalue of the cut-off frequency also or alternatively e.g. includeswitching from a first order to a second order) of the maximum allowedgain setting.

FIG. 9 illustrates one type of embodiments of the invention, FIG. 10 isdirected to a different type of embodiments.

Basically these embodiments comprise a number of elements or steps:

-   1. an AVL (Automatic Volume Leveler): the AVL is a signal dependant    processing block, keeping the volume of the incoming signal at an    approximately constant level,-   2. a (first order) HP (high pass) filter: this filter removes a part    of the lower frequencies, creating headroom for amplification-   3. a gain: increasing the SPL (Sound Pressure Level) of the signal,    i.e. amplifying the signal-   4. a clipper in preferred embodiments, preferably a hard clipper    when a simple system is preferred: the signal is clipped at a    certain amplitude, to assure linear operation of the analogue    amplifier (after D/A conversion). Instead of a hard clipper, which    simply clips the signal above the clipping level, a soft clipper may    also be used, which clips the signal above a clipping level but also    attenuates the signal at level close to the clipping level. Using a    soft-clipper restores to some extent the dynamic behavior of the    signal, increasing intelligibility.-   5. a LP (low pass) filter: the filter restores or at least improves    the balance between mid-range frequencies and high frequencies, an    unbalance makes the sound unnatural, and the signal sounds rather    harsh; this LP filter makes the processed sound more pleasant to    listen to. The cut-off frequencies of the HP and LP filters are    matched.

In this example the input is a speech input, but it is remarked that theinput may be any sound signal.

An exemplary AVL behavior is also shown in FIG. 11.

The left graph shows the step change of the amplitude of three inputsignals. The right graph shows that the AVL gain increases faster forlarge changes in amplitude. This is a preferred embodiment furtherimproving intelligibility.

FIG. 12 illustrates an example of a device similar to the one shown inFIG. 10 in which a sensor (measuring system) 130 for measuringbackground noise, such as e.g. a separate microphone, is used. Thedevice comprises an establisher 131 for the establishing of the cut-offfrequency of the HP-Filter. It is remarked that, although not shown,other parameters, such as the gain, the clipping, the settings of theAVL may be made dependent on the measured noise level.

The noise measurement may, in preferred simple embodiments, give asingle data expressing the overall noise S or may give a noise figuresfor different noise band S_(f1), S_(f2), S_(f3) etc. If noise figuresare measured for different noise bands an average or total noise may becalculated for instance S_(av)=ΣS_(fi), or weighted according to a dB(A)scale for instance S_(av)=Σw_(i)S_(fi) where w_(i) are weightingcoefficients of the dB(A) scale. The noise level is measured by anamplitude measurement.

The HP cut-off frequency is adapted to the measured noise. The higherthe noise level the larger the cut-off frequency. Foradaptive-embodiments the cut-off frequency may advantageously range overa broader range than for non-adaptive embodiments. The HP cut-offfrequency f advantageously range between 50 Hz (for situations in whichthere is substantially no noise) to typically up to 2 kHz forhigh-noise-level situations.

The HP filter cut-off frequency is thus updated according to the amountof environmental noise, and ranges typically from a very low value e.g.50 Hz (no environmental noise) and 2 kHz (loud environmental noise).More low frequencies are removed in loud noisy environments to createmore headroom to amplify the signal afterwards. A maximum of 2 kHz isrecommended to avoid removal of frequencies contributing to the speechintelligibility. The filter coefficients are calculated at run-time.

The relation between cut-off frequency to noise level is preferably setas follows: f_(cut-off)=f₀+Δf(S) where f₀ is the low noise limit (e.g.50, 100 or 300 Hz) and Δf is a higher than linear (proportional to S^(i)where i is greater than 1) function of the noise level.

The matcher 26 matches the LP cut-off frequency to the established HPcut-off frequency, i.e. f′ is set by f′=F(f).

In the absence of noise the cut-off frequency is set at the high limite.g. f_(s)/2.

For maximum noise the cut-off frequency is set at the lower limit, e.g.2 kHz.

FIG. 12 schematically illustrates that the matching itself may bedependent on the gain level, and on the sample frequency fs. Inpreferred embodiments the matcher may operate two ways, i.e. dependingon the sample frequency and the measured noise the LP cut-off frequencyis matched to the HP cut-off frequency or the other way around.

In these embodiments the algorithm used is designed to operateadaptively, driven by the amount of environment (near-end) noise. Thisresults in a user-friendly system feature that allows the system user touse its system (e.g. a GSM) in varying conditions concerningenvironmental noise, without the need of any further interaction tocontrol the GSM volume level.

When used in adaptive processing mode, the parameters of the processingblocks are adapted for incoming samples according to the environmentalnoise. The algorithm adapts parameters according to the environmentalnoise. The amount of noise may be measured by a separate microphone orestimated using the system (GSM) microphone (for a single microphoneapplication).

When the environmental noise decreases in volume the parameters arepreferably adapted very quickly such that the naturalness and warmth ofthe incoming signal are restored.

The term ‘no environmental noise’ does not mean complete silence, butregular noises such as fan noise, background music, etc. In a typicalenvironment, the background noise is typically around 50 dB(A). The term‘loud environmental noise’ refers to the noise of a passing train orsubway, noise inside a dance club, etc. These noises can measure up to100 dB(A).

The noise may be measured by measuring spectral amplitude information ofthe environmental noise and calculating a single value representing theamount of noise.

The relation between the cut-off frequency and the measured noise (ormore in general any parameter on which the cut-off frequency isdependent) is not necessarily linear.

Tests using linear interpolation showed that for ‘medium’ environmentalnoises, the algorithm effect was too large.

Using a higher, for instance second or third order interpolation, theeffect is smaller compared to linear interpolation for the sameenvironmental noise. For loud environmental noises, the amount of effectis equal.

The system in preferred embodiments comprises an adaptor for adapting inaddition to the cut-off frequency of the HP filter one more ofadditional parameters in dependence on the measured background noiselevel. Such parameters are e.g. the order of the high-pass filter.

In short the invention can be described as follows:

A sound reproduction system comprising a digital audio signal input (1),a digital audio signal processor (2, DSP) and a digital audio signaloutput (3) wherein the digital signal processor (2, DSP) comprises ahigh pass (HP) filter (21) with a high pass frequency (f), an amplifier(22) for a signal filtered by the HP filter, and a low pass (LP) filter(23) with a low pass frequency (f′) for filtering the signal afteramplification by the amplifier (22) and for providing an output signal,and the digital processor comprises an establisher (24, 25) forestablishing the high pass frequency or the low pass frequency and amatcher (26) for matching the high pass frequency and low pass frequencyof the high pass filter and low pass filter respectively to each other.

Matching of the HP and LP cut-off frequencies, matches the effect ofthese filters on the perceived speech. In particular it reduces ashrillness of sound which would otherwise be heard.

The invention may be used in various systems. The invention is inparticular useful for hands free mobile phones. However, it isapplicable for all sound reproduction systems, especially those whichrun on a system with a limited voltage supply and/or small loudspeaker.A list of possible applications:

-   -   handsets (mobile phone, DECT, etc.);    -   portable systems, e.g. a portable DVD player    -   PDA'S;    -   car-kits    -   TV's; computers;    -   web-terminals;    -   answering machines;

It will be appreciated by persons skilled in the art that the presentinvention is not limited by what has been particularly shown anddescribed hereinabove. The invention resides in each and every novelcharacteristic feature and each and every combination of characteristicfeatures. Reference numerals in the claims do not limit their protectivescope. Use of the verb “to comprise” and its conjugations does notexclude the presence of elements other than those stated in the claims.Use of the article “a” or “an” preceding an element does not exclude thepresence of a plurality of such elements.

The present invention has been described in terms of specificembodiments, which are illustrative of the invention and not to beconstrued as limiting. The invention may be implemented in hardware,firmware or software, or in a combination of them. Other embodiments arewithin the scope of the following claims.

Many further variations are possible within the concept of theinvention.

1. A sound reproduction system comprising: a digital audio signal input;a digital audio signal processor; a digital audio signal output, asensor that measures a background noise level, and an element having asan input the measured noise level and as an output a HP cut-offfrequency, wherein the HP cut-off frequency increases as the backgroundnoise level increases, and a LP cut-off frequency decreases as the HPcut-off frequency increases, and wherein the digital audio signalprocessor comprises: a high pass (HP) filter, with a HP frequency (f),that filters a signal, an amplifier that amplifies the filtered signal,a low pass (LP) filter, with a LP frequency (f), that filters theamplified signal and provides an output signal, an establisher thatestablishes either the HP frequency or the LP frequency, and a matcherthat matches the HP frequency and the LP frequency of the HP filter andthe LP filter respectively to each other.
 2. A sound reproduction systemas claimed in claim 1, further comprising: a single LP filter with avariable cut-off frequency.
 3. A sound reproduction system as claimed inclaim 1, further comprising: a set of LP filters with a different LPcut-off frequency, wherein the matcher is arranged to send the signalafter amplification to one of the set of LP filters, in dependence onthe HP cut-off frequency.
 4. A sound reproduction system as claimed inclaim 1, wherein the establisher is arranged for establishing thecut-off frequency of the HP filter in dependence on the averageamplification in the amplification stage.
 5. A sound reproduction systemas claimed in claim 1, wherein the establisher is arranged to set thecut-off frequency f′ of the LP filter at f_(s)/2, wherein f_(s) is asample frequency and the matcher matches the HP frequency f to the LPfrequency f′.
 6. A sound reproduction system as claimed in claim 5,further comprising: a single HP filter with a variable cut-offfrequency.
 7. A sound reproduction system as claimed in claim 5, furthercomprising: a set of HP filters with a different HP cut-off frequency,wherein the matcher is arranged to send the signal before amplificationto one of the set of HP filters, in dependence on the LP cut-offfrequency.
 8. A sound reproduction system as claimed in claim 1, whereinthe HP cut-off frequency (f) is a frequency between 300 Hz and 2 kHz. 9.A sound reproduction system as claimed in claim 1, wherein the LPcut-off frequency lies above 2 kHz and f_(s)/2, where f_(s) is a samplefrequency.
 10. A digital audio signal processor comprising: a high pass(HP) filter, with a HP frequency (f), that filters a signal; anamplifier that amplifies the filtered signal; and a low pass (LP)filters, with a LP frequency (f′), that filters the amplified signal andprovides an output signal; an establisher that establishes either the HPfrequency or the LP frequency; a matcher that matches the HP frequencyand the LP frequency respectively to each other; and an element havingas an input a measured noise level from a sensor, the sensor measuring abackground noise level, and, as an output, a HP cut-off frequency,wherein the HP cut-off frequency increases as the background noise levelincreases, and a LP cut-off frequency decreases as the HP cut-offfrequency increases.
 11. A method for processing digital sound signalsin a digital audio signal processor, the method comprising: using a highpass filter to remove frequency components below a HP cut-off frequencyf, thereby producing a filtered signal in the digital audio signalprocessor; using an amplifier to amplify the filtered signal, therebyproducing an amplified signal; using a low pass filter to removefrequency components above a LP cut-off frequency from the amplifiedsignal; using a matcher to match the values of the HP cut-off frequencyand the LP cut-off frequency f′; using a sensor to measure a backgroundnoise level; and adapting a HP cut-off frequency to the measured noiselevel, wherein the HP cut-off frequency increases as the backgroundnoise level increases, and a LP cut-off frequency decreases as the HPcut-off frequency increases.
 12. A method as claimed in claim 11,wherein the HP cut-off frequency lies between 300 and 2 kHz.